1. Field of Invention
The present invention relates generally to the field of telecommunication systems. More particularly the present invention relates to the field of reducing data quality degradation due to encoding/decoding.
2. Background of the Invention
One of the key issues in wireless communications is quality of the service. For voice communications, one measure of quality is the performance of the speech handling systems. An ideal wireless system provides a communications path that is noise free and has high fidelity of reproduction of speech and music. Additionally, because of the preponderance of voice band data applications using modems, the same wireless communications path should ideally support the voice band modems in use on the wireline network.
Unfortunately, this ideal cannot be obtained in commercially practical wireless communications systems that must balance cost and capacity against superb audio quality. Given that the service offered in traditional mobile telephony systems is to enable effective voice communications mainly carrying speech, the wireless speech transport mechanisms purposefully fall short of the ideal.
In analog mobile systems the radio channel bandwidth allocation allows for a speech system, that using FM modulation, can transport a speech band from about 300 Hz to about 3,300 Hz. This is sufficient for a reasonably high fidelity communications system that handles speech, xe2x80x9cmusic-on-hold,xe2x80x9d and medium speed modem data. The analog system is good enough in its fidelity and reproduction capability that multiple cascaded analog connections produce negligible degradation. In fact, prior to the vast digitalization that has taken place, wireline telephony service providers using analog communications circuits could carry the same 300 to 3,300 Hz communications channel across continents and oceans while essentially retaining the quality. The most significant degradation in the analog systems is accumulated noise. This is the pops, crackles, and other perturbations that one traditionally notices.
The advent of digital electronics has changed the nature of the communications channel. While a digital signal does not suffer from the accumulated noise impacts and can remain virtually pure no matter how far it is transported, there is a weak link in the chain of quality. To obtain a digital signal, the analog speech must be converted to a digital form. This conversion process, within the practical constraints of cost effective technology, introduces impairments (i.e., degradation) in the 300 to 3,330 Hz speech channel.
The xe2x80x9chigh endxe2x80x9d of digital telephony is considered to be the conversion of the analog to a digital signals at a rate of 64 kbps. As these signals are transported, there arises the need to convert the signals back to analog form. Often the signal is converted back to digital again. Each analog to digital conversion (and its counterpart digital to analog conversion) adds an additional amount of impairment to the original signal. In the case of the 64 kbps digital signal, approximately 8 tandem analog/digital conversions can be tolerated before the quality is reduced to unacceptable levels.
In mobile communications, the driver for digital telephony has been increased capacity. To achieve additional capacity in the same channel bandwidth allocations previously used by analog FM systems, it is necessary to use an analog to digital conversion technique that encodes the speech at a rate much less than 64 kbps.
As the number of bits per second is reduced, the impairments introduced by the analog to digital conversion and coding process become increasingly large. As encoding rates are reduced, the susceptibility of the payload to impairments in the transmission medium increases. Each information bit becomes more important since it now represents a larger fraction of the desirable information payload. Thus, degradation over a fixed bit error rate channel will increase with lower bit rate encoding schemes. For example, at an 8 kbps coding rate, more than one set of encoding and decoding leads to significant problems. At a 4 kbps coding rate, more than two sets of encoding and decoding produces a virtually unusable communications path.
The issues are much the same for wireless mobile data (as opposed to wireless LANs). While certain quality requirements are different for data than voice, i.e., moderate delay is acceptable for data, but corruption of bit order or loss of bits is totally unacceptable, the issues described above are equally applicable. Data interworking may require rate adaptation, protocol conversion, error correction, etc. Often the interworking of disparate data networks requires treatment of the physical layer and also adaptation of the content, up to and including the presentation layer. This implies that one may need to address all seven layers of the traditional OSI data model for data gateways, a quantum leap from voice interworking which would generally only need treatment of the first two (or possibly three) layers of the OSI model.
The term xe2x80x9ctransliteration,xe2x80x9d as used herein means transforming a signal from one type of coding to another different type of coding.
The term xe2x80x9cvocoderxe2x80x9d as used herein means a voice codec as is commonly used in telephony networks to convert analog voice data to digital data representative of the analog speech and digital-to-analog conversions on digital data representative of analog voice data to the analog data according to predetermined algorithms. As is well-known in the art vocoder algorithms differ in complexity and effective bit rate to achieve varying levels of quality of the voice data as it is subjected to conversions.
The present invention provides architectures for using vocoders that are designed to improve the quality of the speech that traverses through the architecture.
A first preferred embodiment of the present invention is a modified xe2x80x9cbypassxe2x80x9d mode, in which the data is xe2x80x9cmassagedxe2x80x9d prior to being sent. In conventional vocoder bypass, digital voice data can be sent through the base station and mobile switching center (xe2x80x9cMSCxe2x80x9d) and any intervening network elements without modification. However, the intervening network might impair the data in some way. According to the present invention, the data is massaged prior to being sent through the intervening network to mitigate the effect of this impairment on the data.
A second preferred embodiment of the present invention is a xe2x80x9ccommon inter-working facilityxe2x80x9d mode. According to the second preferred embodiment of the present invention, a xe2x80x9cstandardxe2x80x9d vocoder format is defined. Prior to transmitting voice data to the receiving subscriber unit""s MSC, it is converted to the standard format. The data is sent to the receiving mobile unit of the receiving subscriber unit""s MSC, for conversion to whatever vocoder format the receiving subscriber unit normally uses. If the conversion is performed by the subscriber units, this embodiment can be combined with vocoder bypass to avoid conversions in the MSC. The standard format can be any arbitrary vocoder format.
In a third preferred embodiment of the present invention, vocoder xe2x80x9cimpersonationxe2x80x9d is used. In this embodiment, the digital voice data is converted to the receiving subscriber unit""s vocoder format. The converted data is then sent to the receiving subscriber unit. If the conversion is performed by the sending subscriber unit, vocoder impersonation can be combined with vocoder bypass to avoid conversions in the MSC.
A fourth preferred embodiment of the present invention uses vocoder xe2x80x9csubstitution.xe2x80x9d In vocoder substitution, a vocoder format is selected. The selected vocoder format must be available in each of elements that the voice data passes through, specifically, the sending subscriber unit, the receiving subscriber unit, and their MSCs. The data is converted to the selected format and sent to the receiving mobile unit. Where the subscriber units perform the conversions, vocoder bypass can be used to avoid any conversions in the MSC.
Which format to use depends on many factors, including which vocoder formats are being used, desired speech quality and processing capability of the subscriber units. The present invention provides for communication between the sending and receiving subscriber units to determine which vocoder format to use. In the preferred embodiment, this is done through messaging using the SS7 intelligent network associated with mobile telephony. Messages are sent between the sending and receiving subscriber units to determine which vocoder format to use. In some cases, that format may not be available to one or the other of the subscriber units. In that case, the required vocoder can be downloaded from a vocoder storage area. Alternatively, the decision can be made to perform all vocoding functions in the base station or MSC and not in the subscriber units.
Another consideration affecting data quality is the particular route the data takes through the intervening network. For example, when using bypass, any non-conforming element in the intervening network will require additional decoding and encoding steps. As described above, these steps degrade the quality of the underlying transmitted voice signal. Using the common channel signaling provided by SS7, the MSC can assign intervening network elements to handle the call. That is, the present invention can configure the intervening network to minimize impairments to the underlying voice data being transmitted. Another configuration consideration is tandem order. Where cascaded encoding/decoding is required, the order is chosen so that the highest quality encoding/decodings are performed first.
Further, the present invention describes the concept of a universal decoder. The universal decoder is preferably software or hardware configurable to implement any vocoder format. The universal vocoder is can be used to convert voice data to any desired vocoder format. In addition, in the receiving subscriber unit, the universal decoder can automatically determine the correct vocoder format. This can be done in a number of ways including a brute force method in which the incoming voice data is decoded against all vocoder formats in the universal decoder, and the best match is chosen. Preferably, the match is based on frame structure and error functions.
Thus, one object of the present invention is to reduce or eliminate the degradation of voice quality due to encoding and decoding.
Another object of the present invention is to use vocoder substitution to reduce degradation to voice data.
Another object of the present invention is to use vocoder translation to reduce degradation to voice data.
Another object of the present invention is to use vocoder bypass with data massaging to reduce degradation to voice data.
Another object of the present invention is to use vocoder impersonation to reduce degradation to voice data.
Another object of the present invention is to use vocoder substitution to reduce degradation to voice data.
Another object of the present invention is to assign intervening elements over which to route voice data.
Another object of the present invention is to apply a universal vocoder to reduce degradation to voice data.
Another object of the present invention is to reduce degradation when wireline networks communicate with wireless networks.